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430 lines
22 KiB
C++
430 lines
22 KiB
C++
//
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// main.cpp
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// mixer
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//
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// Created by Stephen Birarda on 2/1/13.
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// Copyright (c) 2013 High Fidelity, Inc. All rights reserved.
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//
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#include <iostream>
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#include <math.h>
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#include <string.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <fcntl.h>
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#include <errno.h>
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#include <fstream>
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#include <limits>
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#include <signal.h>
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#include <glm/gtx/norm.hpp>
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#include <glm/gtx/vector_angle.hpp>
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#include <AgentList.h>
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#include <Agent.h>
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#include <AgentTypes.h>
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#include <SharedUtil.h>
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#include <StdDev.h>
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#include <Logstash.h>
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#include "InjectedAudioRingBuffer.h"
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#include "AvatarAudioRingBuffer.h"
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#include <AudioRingBuffer.h>
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#include "PacketHeaders.h"
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#ifdef _WIN32
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#include "Syssocket.h"
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#include "Systime.h"
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#include <math.h>
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#else
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#include <sys/time.h>
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#include <sys/socket.h>
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#include <netinet/in.h>
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#include <arpa/inet.h>
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#endif //_WIN32
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const unsigned short MIXER_LISTEN_PORT = 55443;
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const short JITTER_BUFFER_MSECS = 12;
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const short JITTER_BUFFER_SAMPLES = JITTER_BUFFER_MSECS * (SAMPLE_RATE / 1000.0);
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const long long BUFFER_SEND_INTERVAL_USECS = floorf((BUFFER_LENGTH_SAMPLES_PER_CHANNEL / SAMPLE_RATE) * 1000000);
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const long MAX_SAMPLE_VALUE = std::numeric_limits<int16_t>::max();
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const long MIN_SAMPLE_VALUE = std::numeric_limits<int16_t>::min();
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void plateauAdditionOfSamples(int16_t &mixSample, int16_t sampleToAdd) {
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long sumSample = sampleToAdd + mixSample;
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long normalizedSample = std::min(MAX_SAMPLE_VALUE, sumSample);
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normalizedSample = std::max(MIN_SAMPLE_VALUE, sumSample);
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mixSample = normalizedSample;
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}
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void attachNewBufferToAgent(Agent *newAgent) {
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if (!newAgent->getLinkedData()) {
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if (newAgent->getType() == AGENT_TYPE_AVATAR) {
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newAgent->setLinkedData(new AvatarAudioRingBuffer());
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} else {
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newAgent->setLinkedData(new InjectedAudioRingBuffer());
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}
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}
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}
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bool wantLocalDomain = false;
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int main(int argc, const char* argv[]) {
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setvbuf(stdout, NULL, _IOLBF, 0);
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// Handle Local Domain testing with the --local command line
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const char* local = "--local";
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::wantLocalDomain = cmdOptionExists(argc, argv,local);
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if (::wantLocalDomain) {
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printf("Local Domain MODE!\n");
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int ip = getLocalAddress();
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sprintf(DOMAIN_IP,"%d.%d.%d.%d", (ip & 0xFF), ((ip >> 8) & 0xFF),((ip >> 16) & 0xFF), ((ip >> 24) & 0xFF));
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}
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AgentList* agentList = AgentList::createInstance(AGENT_TYPE_AUDIO_MIXER, MIXER_LISTEN_PORT);
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ssize_t receivedBytes = 0;
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agentList->linkedDataCreateCallback = attachNewBufferToAgent;
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agentList->startSilentAgentRemovalThread();
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unsigned char* packetData = new unsigned char[MAX_PACKET_SIZE];
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sockaddr* agentAddress = new sockaddr;
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// make sure our agent socket is non-blocking
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agentList->getAgentSocket()->setBlocking(false);
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int nextFrame = 0;
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timeval startTime;
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unsigned char clientPacket[BUFFER_LENGTH_BYTES_STEREO + sizeof(PACKET_HEADER_MIXED_AUDIO)];
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clientPacket[0] = PACKET_HEADER_MIXED_AUDIO;
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int16_t clientSamples[BUFFER_LENGTH_SAMPLES_PER_CHANNEL * 2] = {};
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gettimeofday(&startTime, NULL);
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timeval lastDomainServerCheckIn = {};
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timeval beginSendTime, endSendTime;
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float sumFrameTimePercentages = 0.0f;
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int numStatCollections = 0;
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stk::StkFrames stkFrameBuffer(BUFFER_LENGTH_SAMPLES_PER_CHANNEL, 1);
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// if we'll be sending stats, call the Logstash::socket() method to make it load the logstash IP outside the loop
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if (Logstash::shouldSendStats()) {
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Logstash::socket();
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}
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while (true) {
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if (Logstash::shouldSendStats()) {
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gettimeofday(&beginSendTime, NULL);
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}
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// send a check in packet to the domain server if DOMAIN_SERVER_CHECK_IN_USECS has elapsed
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if (usecTimestampNow() - usecTimestamp(&lastDomainServerCheckIn) >= DOMAIN_SERVER_CHECK_IN_USECS) {
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gettimeofday(&lastDomainServerCheckIn, NULL);
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AgentList::getInstance()->sendDomainServerCheckIn();
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if (Logstash::shouldSendStats() && numStatCollections > 0) {
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// if we should be sending stats to Logstash send the appropriate average now
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const char MIXER_LOGSTASH_METRIC_NAME[] = "audio-mixer-frame-time-usage";
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// we're sending a floating point percentage with two mandatory numbers after decimal point
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// that could be up to 6 bytes
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const int MIXER_LOGSTASH_PACKET_BYTES = strlen(MIXER_LOGSTASH_METRIC_NAME) + 7;
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char logstashPacket[MIXER_LOGSTASH_PACKET_BYTES];
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float averageFrameTimePercentage = sumFrameTimePercentages / numStatCollections;
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int packetBytes = sprintf(logstashPacket, "%s %.2f", MIXER_LOGSTASH_METRIC_NAME, averageFrameTimePercentage);
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agentList->getAgentSocket()->send(Logstash::socket(), logstashPacket, packetBytes);
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sumFrameTimePercentages = 0.0f;
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numStatCollections = 0;
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}
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}
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for (AgentList::iterator agent = agentList->begin(); agent != agentList->end(); agent++) {
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PositionalAudioRingBuffer* positionalRingBuffer = (PositionalAudioRingBuffer*) agent->getLinkedData();
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if (positionalRingBuffer && positionalRingBuffer->shouldBeAddedToMix(JITTER_BUFFER_SAMPLES)) {
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// this is a ring buffer that is ready to go
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// set its flag so we know to push its buffer when all is said and done
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positionalRingBuffer->setWillBeAddedToMix(true);
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}
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}
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for (AgentList::iterator agent = agentList->begin(); agent != agentList->end(); agent++) {
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const int PHASE_DELAY_AT_90 = 20;
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if (agent->getType() == AGENT_TYPE_AVATAR) {
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AvatarAudioRingBuffer* agentRingBuffer = (AvatarAudioRingBuffer*) agent->getLinkedData();
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// zero out the client mix for this agent
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memset(clientSamples, 0, sizeof(clientSamples));
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for (AgentList::iterator otherAgent = agentList->begin(); otherAgent != agentList->end(); otherAgent++) {
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if (((PositionalAudioRingBuffer*) otherAgent->getLinkedData())->willBeAddedToMix()
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&& (otherAgent != agent || (otherAgent == agent && agentRingBuffer->shouldLoopbackForAgent()))) {
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PositionalAudioRingBuffer* otherAgentBuffer = (PositionalAudioRingBuffer*) otherAgent->getLinkedData();
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float bearingRelativeAngleToSource = 0.0f;
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float attenuationCoefficient = 1.0f;
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int numSamplesDelay = 0;
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float weakChannelAmplitudeRatio = 1.0f;
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stk::TwoPole* otherAgentTwoPole = NULL;
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if (otherAgent != agent) {
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glm::vec3 listenerPosition = agentRingBuffer->getPosition();
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glm::vec3 relativePosition = otherAgentBuffer->getPosition() - agentRingBuffer->getPosition();
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glm::quat inverseOrientation = glm::inverse(agentRingBuffer->getOrientation());
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float distanceSquareToSource = glm::dot(relativePosition, relativePosition);
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float radius = 0.0f;
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if (otherAgent->getType() == AGENT_TYPE_AUDIO_INJECTOR) {
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InjectedAudioRingBuffer* injectedBuffer = (InjectedAudioRingBuffer*) otherAgentBuffer;
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radius = injectedBuffer->getRadius();
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attenuationCoefficient *= injectedBuffer->getAttenuationRatio();
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}
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if (radius == 0 || (distanceSquareToSource > radius * radius)) {
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// this is either not a spherical source, or the listener is outside the sphere
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if (radius > 0) {
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// this is a spherical source - the distance used for the coefficient
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// needs to be the closest point on the boundary to the source
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// ovveride the distance to the agent with the distance to the point on the
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// boundary of the sphere
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distanceSquareToSource -= (radius * radius);
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} else {
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// calculate the angle delivery for off-axis attenuation
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glm::vec3 rotatedListenerPosition = glm::inverse(otherAgentBuffer->getOrientation())
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* relativePosition;
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float angleOfDelivery = glm::angle(glm::vec3(0.0f, 0.0f, -1.0f),
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glm::normalize(rotatedListenerPosition));
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const float MAX_OFF_AXIS_ATTENUATION = 0.2f;
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const float OFF_AXIS_ATTENUATION_FORMULA_STEP = (1 - MAX_OFF_AXIS_ATTENUATION) / 2.0f;
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float offAxisCoefficient = MAX_OFF_AXIS_ATTENUATION +
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(OFF_AXIS_ATTENUATION_FORMULA_STEP * (angleOfDelivery / 90.0f));
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// multiply the current attenuation coefficient by the calculated off axis coefficient
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attenuationCoefficient *= offAxisCoefficient;
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}
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glm::vec3 rotatedSourcePosition = inverseOrientation * relativePosition;
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const float DISTANCE_SCALE = 2.5f;
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const float GEOMETRIC_AMPLITUDE_SCALAR = 0.3f;
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const float DISTANCE_LOG_BASE = 2.5f;
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const float DISTANCE_SCALE_LOG = logf(DISTANCE_SCALE) / logf(DISTANCE_LOG_BASE);
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// calculate the distance coefficient using the distance to this agent
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float distanceCoefficient = powf(GEOMETRIC_AMPLITUDE_SCALAR,
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DISTANCE_SCALE_LOG +
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(0.5f * logf(distanceSquareToSource) / logf(DISTANCE_LOG_BASE)) - 1);
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distanceCoefficient = std::min(1.0f, distanceCoefficient);
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// multiply the current attenuation coefficient by the distance coefficient
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attenuationCoefficient *= distanceCoefficient;
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// project the rotated source position vector onto the XZ plane
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rotatedSourcePosition.y = 0.0f;
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// produce an oriented angle about the y-axis
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bearingRelativeAngleToSource = glm::orientedAngle(glm::vec3(0.0f, 0.0f, -1.0f),
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glm::normalize(rotatedSourcePosition),
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glm::vec3(0.0f, 1.0f, 0.0f));
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const float PHASE_AMPLITUDE_RATIO_AT_90 = 0.5;
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// figure out the number of samples of delay and the ratio of the amplitude
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// in the weak channel for audio spatialization
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float sinRatio = fabsf(sinf(glm::radians(bearingRelativeAngleToSource)));
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numSamplesDelay = PHASE_DELAY_AT_90 * sinRatio;
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weakChannelAmplitudeRatio = 1 - (PHASE_AMPLITUDE_RATIO_AT_90 * sinRatio);
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// grab the TwoPole object for this source, add it if it doesn't exist
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TwoPoleAgentMap& agentTwoPoles = agentRingBuffer->getTwoPoles();
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TwoPoleAgentMap::iterator twoPoleIterator = agentTwoPoles.find(otherAgent->getAgentID());
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if (twoPoleIterator == agentTwoPoles.end()) {
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// setup the freeVerb effect for this source for this client
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otherAgentTwoPole = agentTwoPoles[otherAgent->getAgentID()] = new stk::TwoPole;
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} else {
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otherAgentTwoPole = twoPoleIterator->second;
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}
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// calculate the reasonance for this TwoPole based on angle to source
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float TWO_POLE_CUT_OFF_FREQUENCY = 800.0f;
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float TWO_POLE_MAX_FILTER_STRENGTH = 0.4f;
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otherAgentTwoPole->setResonance(TWO_POLE_CUT_OFF_FREQUENCY,
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TWO_POLE_MAX_FILTER_STRENGTH
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* fabsf(bearingRelativeAngleToSource) / 180.0f,
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true);
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}
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}
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int16_t* sourceBuffer = otherAgentBuffer->getNextOutput();
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int16_t* goodChannel = (bearingRelativeAngleToSource > 0.0f)
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? clientSamples
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: clientSamples + BUFFER_LENGTH_SAMPLES_PER_CHANNEL;
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int16_t* delayedChannel = (bearingRelativeAngleToSource > 0.0f)
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? clientSamples + BUFFER_LENGTH_SAMPLES_PER_CHANNEL
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: clientSamples;
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int16_t* delaySamplePointer = otherAgentBuffer->getNextOutput() == otherAgentBuffer->getBuffer()
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? otherAgentBuffer->getBuffer() + RING_BUFFER_LENGTH_SAMPLES - numSamplesDelay
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: otherAgentBuffer->getNextOutput() - numSamplesDelay;
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for (int s = 0; s < BUFFER_LENGTH_SAMPLES_PER_CHANNEL; s++) {
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// load up the stkFrameBuffer with this source's samples
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stkFrameBuffer[s] = (stk::StkFloat) sourceBuffer[s];
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}
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// perform the TwoPole effect on the stkFrameBuffer
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if (otherAgentTwoPole) {
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otherAgentTwoPole->tick(stkFrameBuffer);
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}
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for (int s = 0; s < BUFFER_LENGTH_SAMPLES_PER_CHANNEL; s++) {
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if (s < numSamplesDelay) {
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// pull the earlier sample for the delayed channel
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int earlierSample = delaySamplePointer[s] * attenuationCoefficient * weakChannelAmplitudeRatio;
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plateauAdditionOfSamples(delayedChannel[s], earlierSample);
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}
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int16_t currentSample = stkFrameBuffer[s] * attenuationCoefficient;
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plateauAdditionOfSamples(goodChannel[s], currentSample);
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if (s + numSamplesDelay < BUFFER_LENGTH_SAMPLES_PER_CHANNEL) {
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plateauAdditionOfSamples(delayedChannel[s + numSamplesDelay],
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currentSample * weakChannelAmplitudeRatio);
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}
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if (s >= BUFFER_LENGTH_SAMPLES_PER_CHANNEL - PHASE_DELAY_AT_90) {
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// this could be a delayed sample on the next pass
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// so store the affected back in the ARB
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otherAgentBuffer->getNextOutput()[s] = (int16_t) stkFrameBuffer[s];
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}
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}
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}
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}
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memcpy(clientPacket + sizeof(PACKET_HEADER_MIXED_AUDIO), clientSamples, sizeof(clientSamples));
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agentList->getAgentSocket()->send(agent->getPublicSocket(), clientPacket, sizeof(clientPacket));
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}
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}
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// push forward the next output pointers for any audio buffers we used
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for (AgentList::iterator agent = agentList->begin(); agent != agentList->end(); agent++) {
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PositionalAudioRingBuffer* agentBuffer = (PositionalAudioRingBuffer*) agent->getLinkedData();
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if (agentBuffer && agentBuffer->willBeAddedToMix()) {
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agentBuffer->setNextOutput(agentBuffer->getNextOutput() + BUFFER_LENGTH_SAMPLES_PER_CHANNEL);
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if (agentBuffer->getNextOutput() >= agentBuffer->getBuffer() + RING_BUFFER_LENGTH_SAMPLES) {
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agentBuffer->setNextOutput(agentBuffer->getBuffer());
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}
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agentBuffer->setWillBeAddedToMix(false);
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}
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}
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// pull any new audio data from agents off of the network stack
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while (agentList->getAgentSocket()->receive(agentAddress, packetData, &receivedBytes)) {
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if (packetData[0] == PACKET_HEADER_MICROPHONE_AUDIO_NO_ECHO ||
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packetData[0] == PACKET_HEADER_MICROPHONE_AUDIO_WITH_ECHO) {
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Agent* avatarAgent = agentList->addOrUpdateAgent(agentAddress,
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agentAddress,
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AGENT_TYPE_AVATAR,
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agentList->getLastAgentID());
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if (avatarAgent->getAgentID() == agentList->getLastAgentID()) {
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agentList->increaseAgentID();
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}
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agentList->updateAgentWithData(agentAddress, packetData, receivedBytes);
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if (std::isnan(((PositionalAudioRingBuffer *)avatarAgent->getLinkedData())->getOrientation().x)) {
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// kill off this agent - temporary solution to mixer crash on mac sleep
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avatarAgent->setAlive(false);
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}
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} else if (packetData[0] == PACKET_HEADER_INJECT_AUDIO) {
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Agent* matchingInjector = NULL;
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for (AgentList::iterator agent = agentList->begin(); agent != agentList->end(); agent++) {
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if (agent->getLinkedData()) {
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InjectedAudioRingBuffer* ringBuffer = (InjectedAudioRingBuffer*) agent->getLinkedData();
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if (memcmp(ringBuffer->getStreamIdentifier(),
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packetData + 1,
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STREAM_IDENTIFIER_NUM_BYTES) == 0) {
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// this is the matching stream, assign to matchingInjector and stop looking
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matchingInjector = &*agent;
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break;
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}
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}
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}
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if (!matchingInjector) {
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matchingInjector = agentList->addOrUpdateAgent(NULL,
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NULL,
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AGENT_TYPE_AUDIO_INJECTOR,
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agentList->getLastAgentID());
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agentList->increaseAgentID();
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}
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// give the new audio data to the matching injector agent
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agentList->updateAgentWithData(matchingInjector, packetData, receivedBytes);
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}
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}
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if (Logstash::shouldSendStats()) {
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// send a packet to our logstash instance
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// calculate the percentage value for time elapsed for this send (of the max allowable time)
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gettimeofday(&endSendTime, NULL);
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float percentageOfMaxElapsed = ((float) (usecTimestamp(&endSendTime) - usecTimestamp(&beginSendTime))
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/ BUFFER_SEND_INTERVAL_USECS) * 100.0f;
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if (percentageOfMaxElapsed > 0) {
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sumFrameTimePercentages += percentageOfMaxElapsed;
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}
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numStatCollections++;
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}
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long long usecToSleep = usecTimestamp(&startTime) + (++nextFrame * BUFFER_SEND_INTERVAL_USECS) - usecTimestampNow();
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if (usecToSleep > 0) {
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usleep(usecToSleep);
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} else {
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std::cout << "Took too much time, not sleeping!\n";
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}
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}
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return 0;
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}
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